Communication resources such as radio frequency channels are ultimately limited in capacity. Notwithstanding this limitation, communication needs continue to rapidly increase. Dispatch, selective call, and cellular communications, to name a few, are all being utilized by an increasing number of users. Without appropriate technological advances, many users will face either impaired service or possibly a complete lack of available service.
One recent technological advance intended to increase the efficiency of data throughput, and hence decrease system capacity needs to thereby allow more communications to be supported by the available limited resources, is speech coding. For example, code excited linear prediction speech coders and vector sum excited linear prediction speech coders have been proposed that exhibit good performance at relatively low data rates. Rather than transmitting the original voice information itself, or a digitized version thereof, such speech coders utilize techniques to allow a coded representation of the voice information to be transmitted instead. Utilizing the coded representation upon receipt, the voice message can then be reconstructed. For a general description of one version of such a device, see U.S. Pat. No. 4,933,957 to Bottau et al., which describes a low bit rate voice coding method and system.
In code excited linear prediction (CELP) type coders, long term predictor delay (lag) constitutes one of the parameters that requires encoding. This lag indicates a delay at which long term signal correlation is typically maximized. The various coder parameters and other information are typically transmitted in a plurality of subframes at a given frame rate, and in prior art coders, the lag parameter is typically coded independently and updated at the subframe rate. The frame parameters typically include the linear prediction coefficients, whereas the subframe parameters more typically define the excitation. In a CELP type coder, these excitation parameters include the lag factor, the index of the code vector, and various relevant gains.
When representing voiced speech, the lag factor typically exhibits a high degree of correlation from one subframe to another. By coding the lags independently from subframe to subframe, the prior art has failed to properly exploit this high degree of correlation.
Some prior art suggestions have been offered that attempt to exploit this subframe to subframe lag correlation. Pursuant to these suggestions, the lag information is coded more compactly. To date this compactness has been gained at the cost of reduced long term predictor performance. Such prior art methods typically include coding the lag once per frame (usually every 15-30 milliseconds) with a third order long term predictor being used. If the taps of the long term predictor filter are updated at the subframe rate, a lag change of at most two samples can be accommodated over the duration of the frame. An alternate method sends the frame lag, and codes the long term predictor lag deviations relative to the frame lag at each subframe. Yet another method, described in Federal Standard 1016, codes the lag independently at odd subframes, and employs delta-coding relative to the preceding subframe's coded lag at even subframes.
While successfully reducing the number of bits assigned to code the lag information, each of the above described methods results in a compromise to speech quality performance and/or coding efficiency. A need therefore exists for an improved method and apparatus for efficient encoding of long term predictor lag information, where the gained efficiency does not necessarily result in performance degradation. The reduction in bits utilized to represent the lag information could then either support a reduced coding rate speech coder, or the bit savings could be exploited to employ additional error correction for other information contained within the voice coding.